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/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
#include "gtest/gtest.h"
#include "AudioDriftCorrection.h"
#include "AudioGenerator.h"
#include "AudioVerifier.h"
#include "nsContentUtils.h"
using namespace mozilla;
template <class T>
AudioChunk CreateAudioChunk(uint32_t aFrames, uint32_t aChannels,
AudioSampleFormat aSampleFormat);
void testAudioCorrection(int32_t aSourceRate, int32_t aTargetRate,
bool aTestMonoToStereoInput = false) {
const uint32_t frequency = 100;
const PrincipalHandle testPrincipal =
MakePrincipalHandle(nsContentUtils::GetSystemPrincipal());
AudioDriftCorrection ad(aSourceRate, aTargetRate, testPrincipal);
uint8_t numChannels = 1;
AudioGenerator<AudioDataValue> tone(numChannels, aSourceRate, frequency);
AudioVerifier<AudioDataValue> inToneVerifier(aSourceRate, frequency);
AudioVerifier<AudioDataValue> outToneVerifier(aTargetRate, frequency);
// Run for some time: 3 * 5000 = 15000 iterations
for (uint32_t j = 0; j < 3; ++j) {
TrackTime sourceFramesIteration = 0;
TrackTime targetFramesIteration = 0;
// apply some drift (+/- .2%)
const int8_t additionalDriftFrames =
((j % 2 == 0) ? aSourceRate : -aSourceRate) * 2 / 1000;
// If the number of frames before changing channel count (and thereby
// resetting the resampler) is very low, the measured buffering level curve
// may look odd, as each resampler reset will reset the (possibly
// fractional) output frame counter.
const uint32_t numFramesBeforeChangingChannelCount = aSourceRate;
uint32_t numFramesAtCurrentChannelCount = 0;
// 50 seconds, allows for at least 50 correction changes, to stabilize
// on the current drift.
for (uint32_t n = 0; n < 5000; ++n) {
const TrackTime sourceFrames =
(n + 1) * (aSourceRate + additionalDriftFrames) / 100 -
sourceFramesIteration;
const TrackTime targetFrames =
(n + 1) * aTargetRate / 100 - targetFramesIteration;
AudioSegment inSegment;
if (aTestMonoToStereoInput) {
// Create the input (sine tone) of two chunks.
const TrackTime sourceFramesPart1 = std::min<TrackTime>(
sourceFrames, numFramesBeforeChangingChannelCount -
numFramesAtCurrentChannelCount);
tone.Generate(inSegment, sourceFramesPart1);
numFramesAtCurrentChannelCount += sourceFramesPart1;
if (numFramesBeforeChangingChannelCount ==
numFramesAtCurrentChannelCount) {
tone.SetChannelsCount(numChannels = (numChannels % 2) + 1);
numFramesAtCurrentChannelCount = sourceFrames - sourceFramesPart1;
tone.Generate(inSegment, numFramesAtCurrentChannelCount);
}
} else {
// Create the input (sine tone)
tone.Generate(inSegment, sourceFrames);
}
inToneVerifier.AppendData(inSegment);
// Get the output of the correction
AudioSegment outSegment = ad.RequestFrames(inSegment, targetFrames);
EXPECT_EQ(outSegment.GetDuration(), targetFrames);
for (AudioSegment::ConstChunkIterator ci(outSegment); !ci.IsEnded();
ci.Next()) {
EXPECT_EQ(ci->mPrincipalHandle, testPrincipal);
}
outToneVerifier.AppendData(outSegment);
sourceFramesIteration += sourceFrames;
targetFramesIteration += targetFrames;
}
}
// Initial buffering is 50ms, which is then expected to be reduced as the
// drift adaptation stabilizes.
EXPECT_LT(ad.CurrentBuffering(), aSourceRate * 50U / 1000);
// Desired buffering should not go lower than some 130% of the source buffer
// size per-iteration.
EXPECT_GT(ad.CurrentBuffering(), aSourceRate * 10U / 1000);
EXPECT_EQ(ad.NumUnderruns(), 0U);
EXPECT_FLOAT_EQ(inToneVerifier.EstimatedFreq(), tone.mFrequency);
EXPECT_EQ(inToneVerifier.PreSilenceSamples(), 0U);
EXPECT_EQ(inToneVerifier.CountDiscontinuities(), 0U);
EXPECT_NEAR(outToneVerifier.EstimatedFreq(), tone.mFrequency, 1.0f);
// The expected pre-silence is equal to the initial desired buffering (50ms)
// minus what is left after resampling the first input segment.
const auto buffering = media::TimeUnit::FromSeconds(0.05);
const auto sourceStep =
media::TimeUnit(aSourceRate * 1002 / 1000 / 100, aSourceRate);
const auto targetStep = media::TimeUnit(aTargetRate / 100, aTargetRate);
EXPECT_NEAR(static_cast<int64_t>(outToneVerifier.PreSilenceSamples()),
(targetStep + buffering - sourceStep)
.ToBase(aSourceRate)
.ToBase<media::TimeUnit::CeilingPolicy>(aTargetRate)
.ToTicksAtRate(aTargetRate),
1U);
EXPECT_EQ(outToneVerifier.CountDiscontinuities(), 0U);
}
TEST(TestAudioDriftCorrection, Basic)
{
printf("Testing AudioCorrection 48 -> 48\n");
testAudioCorrection(48000, 48000);
printf("Testing AudioCorrection 48 -> 44.1\n");
testAudioCorrection(48000, 44100);
printf("Testing AudioCorrection 44.1 -> 48\n");
testAudioCorrection(44100, 48000);
printf("Testing AudioCorrection 23458 -> 25113\n");
testAudioCorrection(23458, 25113);
}
TEST(TestAudioDriftCorrection, MonoToStereoInput)
{
constexpr bool testMonoToStereoInput = true;
printf("Testing MonoToStereoInput 48 -> 48\n");
testAudioCorrection(48000, 48000, testMonoToStereoInput);
printf("Testing MonoToStereoInput 48 -> 44.1\n");
testAudioCorrection(48000, 44100, testMonoToStereoInput);
printf("Testing MonoToStereoInput 44.1 -> 48\n");
testAudioCorrection(44100, 48000, testMonoToStereoInput);
}
TEST(TestAudioDriftCorrection, NotEnoughFrames)
{
const uint32_t frequency = 100;
const uint32_t sampleRateTransmitter = 48000;
const uint32_t sampleRateReceiver = 48000;
const PrincipalHandle testPrincipal =
MakePrincipalHandle(nsContentUtils::GetSystemPrincipal());
AudioDriftCorrection ad(sampleRateTransmitter, sampleRateReceiver,
testPrincipal);
const uint32_t targetFrames = sampleRateReceiver / 100;
AudioGenerator<AudioDataValue> tone(1, sampleRateTransmitter, frequency);
AudioVerifier<AudioDataValue> outToneVerifier(sampleRateReceiver, frequency);
for (uint32_t i = 0; i < 7; ++i) {
// Input is something small, 10 frames here, in order to dry out fast,
// after 4 iterations (pre-buffer = 2400)
AudioSegment inSegment;
tone.Generate(inSegment, 10);
AudioSegment outSegment = ad.RequestFrames(inSegment, targetFrames);
EXPECT_EQ(outSegment.GetDuration(), targetFrames);
EXPECT_FALSE(outSegment.IsNull());
for (AudioSegment::ConstChunkIterator ci(outSegment); !ci.IsEnded();
ci.Next()) {
if (i < 5) {
if (!ci->IsNull()) {
EXPECT_EQ(ci->mPrincipalHandle, testPrincipal);
}
}
}
outToneVerifier.AppendData(outSegment);
}
EXPECT_EQ(ad.BufferSize(), 4800U);
EXPECT_EQ(ad.NumUnderruns(), 1u);
EXPECT_EQ(outToneVerifier.CountDiscontinuities(), 1u);
}
TEST(TestAudioDriftCorrection, CrashInAudioResampler)
{
const uint32_t sampleRateTransmitter = 48000;
const uint32_t sampleRateReceiver = 48000;
const PrincipalHandle testPrincipal =
MakePrincipalHandle(nsContentUtils::GetSystemPrincipal());
AudioDriftCorrection ad(sampleRateTransmitter, sampleRateReceiver,
testPrincipal);
const uint32_t targetFrames = sampleRateReceiver / 100;
for (uint32_t i = 0; i < 100; ++i) {
AudioChunk chunk = CreateAudioChunk<float>(sampleRateTransmitter / 1000, 1,
AUDIO_FORMAT_FLOAT32);
AudioSegment inSegment;
inSegment.AppendAndConsumeChunk(std::move(chunk));
AudioSegment outSegment = ad.RequestFrames(inSegment, targetFrames);
EXPECT_EQ(outSegment.GetDuration(), targetFrames);
for (AudioSegment::ConstChunkIterator ci(outSegment); !ci.IsEnded();
ci.Next()) {
if (!ci->IsNull()) { // Don't check the data if ad is dried out.
EXPECT_EQ(ci->mPrincipalHandle, testPrincipal);
}
}
}
}
TEST(TestAudioDriftCorrection, HighLatencyProducerLowLatencyConsumer)
{
constexpr uint32_t transmitterBlockSize = 2048;
constexpr uint32_t receiverBlockSize = 128;
constexpr uint32_t sampleRate = 48000;
const PrincipalHandle testPrincipal =
MakePrincipalHandle(nsContentUtils::GetSystemPrincipal());
AudioDriftCorrection ad(sampleRate, sampleRate, testPrincipal);
uint32_t numBlocksProduced = 0;
for (uint32_t i = 0; i < (sampleRate / 1000) * 500; i += receiverBlockSize) {
AudioSegment inSegment;
if ((i / transmitterBlockSize) > numBlocksProduced) {
AudioChunk chunk = CreateAudioChunk<float>(transmitterBlockSize, 1,
AUDIO_FORMAT_FLOAT32);
inSegment.AppendAndConsumeChunk(std::move(chunk));
++numBlocksProduced;
}
AudioSegment outSegment = ad.RequestFrames(inSegment, receiverBlockSize);
EXPECT_EQ(outSegment.GetDuration(), receiverBlockSize);
}
// Input is stable so no corrections should occur.
EXPECT_EQ(ad.NumCorrectionChanges(), 0U);
}
TEST(TestAudioDriftCorrection, LargerTransmitterBlockSizeThanDesiredBuffering)
{
constexpr uint32_t transmitterBlockSize = 4096;
constexpr uint32_t receiverBlockSize = 128;
constexpr uint32_t sampleRate = 48000;
const PrincipalHandle testPrincipal =
MakePrincipalHandle(nsContentUtils::GetSystemPrincipal());
AudioDriftCorrection ad(sampleRate, sampleRate, testPrincipal);
uint32_t numBlocksTransmitted = 0;
for (uint32_t i = 0; i < (sampleRate / 1000) * 500; i += receiverBlockSize) {
AudioSegment inSegment;
if (uint32_t currentBlock = i / transmitterBlockSize;
currentBlock > numBlocksTransmitted) {
AudioChunk chunk = CreateAudioChunk<float>(transmitterBlockSize, 1,
AUDIO_FORMAT_FLOAT32);
inSegment.AppendAndConsumeChunk(std::move(chunk));
numBlocksTransmitted = currentBlock;
}
AudioSegment outSegment = ad.RequestFrames(inSegment, receiverBlockSize);
EXPECT_EQ(outSegment.GetDuration(), receiverBlockSize);
if (numBlocksTransmitted > 0) {
EXPECT_GT(ad.CurrentBuffering(), 0U);
}
}
// Input is stable so no corrections should occur.
EXPECT_EQ(ad.NumCorrectionChanges(), 0U);
// The desired buffering and pre-buffering level was
// transmitterBlockSize * 11 / 10 to accomodate the large input block size.
// The buffer size was twice the pre-buffering level.
EXPECT_EQ(ad.BufferSize(), transmitterBlockSize * 11 / 10 * 2);
}
TEST(TestAudioDriftCorrection, LargerReceiverBlockSizeThanDesiredBuffering)
{
constexpr uint32_t transmitterBlockSize = 128;
constexpr uint32_t receiverBlockSize = 4096;
constexpr uint32_t sampleRate = 48000;
const PrincipalHandle testPrincipal =
MakePrincipalHandle(nsContentUtils::GetSystemPrincipal());
AudioDriftCorrection ad(sampleRate, sampleRate, testPrincipal);
AudioSegment inSegment;
for (uint32_t i = 0; i < (sampleRate / 1000) * 500;
i += transmitterBlockSize) {
AudioChunk chunk =
CreateAudioChunk<float>(transmitterBlockSize, 1, AUDIO_FORMAT_FLOAT32);
inSegment.AppendAndConsumeChunk(std::move(chunk));
if (i % receiverBlockSize == 0) {
AudioSegment outSegment = ad.RequestFrames(inSegment, receiverBlockSize);
EXPECT_EQ(outSegment.GetDuration(), receiverBlockSize);
inSegment.Clear();
}
if (i >= receiverBlockSize) {
EXPECT_GT(ad.CurrentBuffering(), 0U);
}
}
// Input is stable so no corrections should occur.
EXPECT_EQ(ad.NumCorrectionChanges(), 0U);
EXPECT_EQ(ad.NumUnderruns(), 0U);
// The drift correction buffer size had to be larger than the desired (the
// buffer size is twice the initial buffering level), to accomodate the large
// input block size that gets buffered in the resampler only when processing
// output.
EXPECT_EQ(ad.BufferSize(), 9600U);
}
TEST(TestAudioDriftCorrection, DynamicInputBufferSizeChanges)
{
constexpr uint32_t transmitterBlockSize1 = 2048;
constexpr uint32_t transmitterBlockSize2 = 4096;
constexpr uint32_t receiverBlockSize = 128;
constexpr uint32_t sampleRate = 48000;
constexpr uint32_t frequencyHz = 100;
const PrincipalHandle testPrincipal =
MakePrincipalHandle(nsContentUtils::GetSystemPrincipal());
AudioDriftCorrection ad(sampleRate, sampleRate, testPrincipal);
AudioGenerator<AudioDataValue> tone(1, sampleRate, frequencyHz);
AudioVerifier<AudioDataValue> inToneVerifier(sampleRate, frequencyHz);
AudioVerifier<AudioDataValue> outToneVerifier(sampleRate, frequencyHz);
TrackTime totalFramesTransmitted = 0;
TrackTime totalFramesReceived = 0;
const auto produceSomeData = [&](uint32_t aTransmitterBlockSize,
uint32_t aDuration) {
TrackTime transmittedFramesStart = totalFramesTransmitted;
TrackTime receivedFramesStart = totalFramesReceived;
uint32_t numBlocksTransmitted = 0;
for (uint32_t i = 0; i < aDuration; i += receiverBlockSize) {
AudioSegment inSegment;
if (((receivedFramesStart - transmittedFramesStart + i) /
aTransmitterBlockSize) > numBlocksTransmitted) {
tone.Generate(inSegment, aTransmitterBlockSize);
MOZ_RELEASE_ASSERT(!inSegment.IsNull());
inToneVerifier.AppendData(inSegment);
MOZ_RELEASE_ASSERT(!inSegment.IsNull());
++numBlocksTransmitted;
totalFramesTransmitted += aTransmitterBlockSize;
}
AudioSegment outSegment = ad.RequestFrames(inSegment, receiverBlockSize);
EXPECT_EQ(outSegment.GetDuration(), receiverBlockSize);
outToneVerifier.AppendData(outSegment);
totalFramesReceived += receiverBlockSize;
}
};
produceSomeData(transmitterBlockSize1, 5 * sampleRate);
EXPECT_EQ(ad.BufferSize(), 4800U);
// No input is provided for the first transmitterBlockSize1 of output
// requested. This causes a lower input rate estimate, so there are some
// initial corrections.
EXPECT_GT(ad.NumCorrectionChanges(), 0U);
EXPECT_EQ(ad.NumUnderruns(), 0U);
// Increase input latency. We expect this to underrun, but only once as the
// drift correction adapts its buffer size and desired buffering level.
produceSomeData(transmitterBlockSize2, 25 * sampleRate);
auto numCorrectionChanges = ad.NumCorrectionChanges();
EXPECT_EQ(ad.NumUnderruns(), 1U);
// Adapting to the new input block size should have stabilized.
EXPECT_GT(ad.BufferSize(), transmitterBlockSize2);
produceSomeData(transmitterBlockSize2, 10 * sampleRate);
EXPECT_LE(ad.NumCorrectionChanges(), numCorrectionChanges + 1);
EXPECT_EQ(ad.NumUnderruns(), 1U);
// Decrease input latency. We expect the drift correction to gradually
// decrease its desired buffering level.
produceSomeData(transmitterBlockSize1, 100 * sampleRate);
numCorrectionChanges = ad.NumCorrectionChanges();
EXPECT_EQ(ad.NumUnderruns(), 1U);
EXPECT_EQ(ad.BufferSize(), 9600U);
// Adjustments to the desired buffering level continue.
produceSomeData(transmitterBlockSize1, 20 * sampleRate);
EXPECT_LE(ad.NumCorrectionChanges(), numCorrectionChanges + 2);
EXPECT_EQ(ad.NumUnderruns(), 1U);
EXPECT_NEAR(inToneVerifier.EstimatedFreq(), tone.mFrequency, 1.0f);
EXPECT_EQ(inToneVerifier.PreSilenceSamples(), 0U);
EXPECT_EQ(inToneVerifier.CountDiscontinuities(), 0U);
EXPECT_NEAR(outToneVerifier.EstimatedFreq(), tone.mFrequency, 1.0f);
// No input is provided for the first transmitterBlockSize1 of output
// requested, so this output is all silent.
// The expected additional pre-silence is equal to the desired buffering
// plus what's needed to produce the first output segment minus the first
// input segment.
EXPECT_EQ(outToneVerifier.PreSilenceSamples(), 2528U);
EXPECT_NEAR(outToneVerifier.PreSilenceSamples() - transmitterBlockSize1,
media::TimeUnit::FromSeconds(0.05).ToTicksAtRate(sampleRate) +
receiverBlockSize - transmitterBlockSize1,
1U);
// One mid-stream period of silence from increasing the input buffer size,
// causing an underrun. Counts as two discontinuities.
EXPECT_EQ(outToneVerifier.CountDiscontinuities(), 2U);
}
/**
* This is helpful to run together with
* MOZ_LOG=raw,DriftControllerGraphs:5 MOZ_LOG_FILE=./plot_values.csv
* to be able to plot the step response of a change in source clock rate (i.e.
* drift). Useful for calculating and verifying PID coefficients.
*/
TEST(TestAudioDriftCorrection, DriftStepResponse)
{
constexpr uint32_t nominalRate = 48000;
constexpr uint32_t interval = nominalRate;
constexpr uint32_t inputRate = nominalRate * 1005 / 1000; // 0.5% drift
constexpr uint32_t inputInterval = inputRate;
constexpr uint32_t iterations = 200;
const PrincipalHandle testPrincipal =
MakePrincipalHandle(nsContentUtils::GetSystemPrincipal());
AudioGenerator<AudioDataValue> tone(1, nominalRate, 440);
AudioDriftCorrection ad(nominalRate, nominalRate, testPrincipal);
for (uint32_t i = 0; i < interval * iterations; i += interval / 100) {
AudioSegment inSegment;
tone.Generate(inSegment, inputInterval / 100);
ad.RequestFrames(inSegment, interval / 100);
}
EXPECT_EQ(ad.BufferSize(), 4800U);
EXPECT_EQ(ad.NumUnderruns(), 0u);
}
/**
* Similar to DriftStepResponse but will underrun to allow testing the underrun
* handling. This is helpful to run together with
* MOZ_LOG=raw,DriftControllerGraphs:5 MOZ_LOG_FILE=./plot_values.csv
*/
TEST(TestAudioDriftCorrection, DriftStepResponseUnderrun)
{
constexpr uint32_t nominalRate = 48000;
constexpr uint32_t interval = nominalRate;
constexpr uint32_t iterations = 200;
const PrincipalHandle testPrincipal =
MakePrincipalHandle(nsContentUtils::GetSystemPrincipal());
uint32_t inputRate = nominalRate * 1005 / 1000; // 0.5% drift
uint32_t inputInterval = inputRate;
AudioGenerator<AudioDataValue> tone(1, nominalRate, 440);
AudioDriftCorrection ad(nominalRate, nominalRate, testPrincipal);
for (uint32_t i = 0; i < interval * iterations; i += interval / 100) {
AudioSegment inSegment;
tone.Generate(inSegment, inputInterval / 100);
ad.RequestFrames(inSegment, interval / 100);
}
inputRate = nominalRate * 997 / 1000; // -0.3% drift
inputInterval = inputRate;
for (uint32_t i = 0; i < interval * iterations; i += interval / 100) {
AudioSegment inSegment;
tone.Generate(inSegment, inputInterval / 100);
ad.RequestFrames(inSegment, interval / 100);
}
EXPECT_EQ(ad.BufferSize(), 4800U);
EXPECT_EQ(ad.NumUnderruns(), 1u);
}
/**
* Similar to DriftStepResponse but with a high-latency input, and will underrun
* to allow testing the underrun handling. This is helpful to run together with
* MOZ_LOG=raw,DriftControllerGraphs:5 MOZ_LOG_FILE=./plot_values.csv
*/
TEST(TestAudioDriftCorrection, DriftStepResponseUnderrunHighLatencyInput)
{
constexpr uint32_t nominalRate = 48000;
constexpr uint32_t interval = nominalRate;
constexpr uint32_t iterations = 200;
const PrincipalHandle testPrincipal =
MakePrincipalHandle(nsContentUtils::GetSystemPrincipal());
uint32_t inputRate1 = nominalRate * 1005 / 1000; // 0.5% drift
uint32_t inputInterval1 = inputRate1;
AudioGenerator<AudioDataValue> tone(1, nominalRate, 440);
AudioDriftCorrection ad(nominalRate, nominalRate, testPrincipal);
for (uint32_t i = 0; i < interval * iterations; i += interval / 100) {
AudioSegment inSegment;
if (i > 0 && i % interval == 0) {
tone.Generate(inSegment, inputInterval1);
}
ad.RequestFrames(inSegment, interval / 100);
}
uint32_t inputRate2 = nominalRate * 995 / 1000; // -0.5% drift
uint32_t inputInterval2 = inputRate2;
for (uint32_t i = 0; i < interval * iterations; i += interval / 100) {
AudioSegment inSegment;
// The first segment is skipped to cause an underrun.
if (i > 0 && i % interval == 0) {
tone.Generate(inSegment, inputInterval2);
}
ad.RequestFrames(inSegment, interval / 100);
if (i >= interval * 8 / 10 && i < interval) {
// While the DynamicResampler has not set its pre-buffer after the
// underrun, InFramesBuffered() reports the pre-buffer size.
// The initial desired buffer and pre-buffer size was
// inputInterval1 * 11 / 10 to accomodate the large input block size.
// This was doubled when the underrun occurred.
EXPECT_EQ(ad.CurrentBuffering(), inputInterval1 * 11 / 10 * 2)
<< "for i=" << i;
} else if (i == interval) {
// After the pre-buffer was set and used to generate the first output
// with a non-empty input segment after the underun, the actual number
// of frames buffered almost matches the pre-buffer size, with some
// rounding from output to input frame count conversion.
EXPECT_NEAR(ad.CurrentBuffering(), inputInterval1 * 11 / 10 * 2, 1)
<< "after first input after underrun";
}
}
// The initial desired buffering and pre-buffering level was
// inputInterval1 * 11 / 10 to accomodate the large input block size.
// The buffer size was initially twice the pre-buffering level, and then
// doubled when the underrun occurred.
EXPECT_EQ(ad.BufferSize(), inputInterval1 * 11 / 10 * 2 * 2);
EXPECT_EQ(ad.NumUnderruns(), 1u);
}
/**
* Similar to DriftStepResponse but with a high-latency input, and will overrun
* (input callback buffer is larger than AudioDriftCorrection's ring buffer for
* input data) to allow testing the overrun handling. This is helpful to run
* together with
* MOZ_LOG=raw,DriftControllerGraphs:5 MOZ_LOG_FILE=./plot_values.csv
*/
TEST(TestAudioDriftCorrection, DriftStepResponseOverrun)
{
constexpr uint32_t nominalRate = 48000;
constexpr uint32_t interval = nominalRate;
constexpr uint32_t inputRate = nominalRate * 1005 / 1000; // 0.5% drift
constexpr uint32_t inputInterval = inputRate;
constexpr uint32_t iterations = 200;
const PrincipalHandle testPrincipal =
MakePrincipalHandle(nsContentUtils::GetSystemPrincipal());
AudioGenerator<AudioDataValue> tone(1, nominalRate, 440);
AudioDriftCorrection ad(nominalRate, nominalRate, testPrincipal);
for (uint32_t i = 0; i < interval * iterations; i += interval / 100) {
AudioSegment inSegment;
tone.Generate(inSegment, inputInterval / 100);
ad.RequestFrames(inSegment, interval / 100);
}
// Change input callbacks to 1000ms (+0.5% drift) = 48200 frames, which will
// overrun the ring buffer.
for (uint32_t i = 0; i < interval * iterations; i += interval / 100) {
AudioSegment inSegment;
if (i > 0 && i % interval == 0) {
// This simulates the input stream latency increasing externally. It's
// building up a second worth of data before the next callback. This also
// causes an underrun.
tone.Generate(inSegment, inputInterval);
}
ad.RequestFrames(inSegment, interval / 100);
}
// The desired buffering and pre-buffering levels were increased to
// inputInterval * 11 / 10 to accomodate the large input block size.
// The buffer size was increased to twice the pre-buffering level.
EXPECT_EQ(ad.BufferSize(), inputInterval * 11 / 10 * 2);
EXPECT_EQ(ad.NumUnderruns(), 1u);
}